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22 #include "config_components.h"
38 int i, flags1, flags2, block_align;
46 "too many channels: got %i, need %i or fewer\n",
59 "bitrate too low: got %"PRId64
", need 24000 or higher\n",
85 s->use_exp_vlc = flags2 & 0x0001;
86 s->use_bit_reservoir = flags2 & 0x0002;
87 s->use_variable_block_len = flags2 & 0x0004;
95 for (
i = 0;
i <
s->nb_block_sizes;
i++) {
98 0, 1 << (
s->frame_len_bits -
i), &
scale, 0);
115 const float *
const *audio = (
const float *
const *)
frame->extended_data;
117 int window_index =
s->frame_len_bits -
s->block_len_bits;
119 av_tx_fn mdct_fn =
s->mdct_fn[window_index];
121 const float *
win =
s->windows[window_index];
122 int window_len = 1 <<
s->block_len_bits;
123 float n = 2.0 * 32768.0 / window_len;
126 memcpy(
s->output,
s->frame_out[ch], window_len *
sizeof(*
s->output));
127 s->fdsp->vector_fmul_scalar(
s->frame_out[ch], audio[ch], n,
len);
128 s->fdsp->vector_fmul_reverse(&
s->output[window_len],
s->frame_out[ch],
130 s->fdsp->vector_fmul(
s->frame_out[ch],
s->frame_out[ch],
win,
len);
131 mdct_fn(mdct,
s->coefs[ch],
s->output,
sizeof(
float));
146 float v, *q, max_scale, *q_end;
148 ptr =
s->exponent_bands[
s->frame_len_bits -
s->block_len_bits];
149 q =
s->exponents[ch];
150 q_end = q +
s->block_len;
154 v =
ff_exp10(*exp_param++ *(1.0 / 16.0));
155 max_scale =
FFMAX(max_scale, v);
161 s->max_exponent[ch] = max_scale;
170 ptr =
s->exponent_bands[
s->frame_len_bits -
s->block_len_bits];
171 q =
s->exponents[ch];
172 q_end = q +
s->block_len;
173 if (
s->version == 1) {
174 last_exp = *exp_param++;
175 av_assert0(last_exp - 10 >= 0 && last_exp - 10 < 32);
181 int exp = *exp_param++;
182 int code =
exp - last_exp + 60;
195 int channels =
s->avctx->ch_layout.nb_channels;
208 if (
s->use_variable_block_len) {
212 s->next_block_len_bits =
s->frame_len_bits;
213 s->prev_block_len_bits =
s->frame_len_bits;
214 s->block_len_bits =
s->frame_len_bits;
217 s->block_len = 1 <<
s->block_len_bits;
219 bsize =
s->frame_len_bits -
s->block_len_bits;
222 v =
s->coefs_end[bsize] -
s->coefs_start;
226 int n4 =
s->block_len / 2;
227 mdct_norm = 1.0 / (
float) n4;
229 mdct_norm *= sqrt(n4);
237 s->channel_coded[ch] = 1;
238 if (
s->channel_coded[ch])
243 if (
s->channel_coded[ch]) {
245 float *coefs, *exponents,
mult;
248 coefs1 =
s->coefs1[ch];
249 exponents =
s->exponents[ch];
252 coefs = src_coefs[ch];
253 if (
s->use_noise_coding && 0) {
256 coefs +=
s->coefs_start;
258 for (
i = 0;
i < n;
i++) {
259 double t = *coefs++ / (exponents[
i] *
mult);
260 if (t < -32768 || t > 32767)
271 int a =
s->channel_coded[ch];
279 for (v = total_gain - 1; v >= 127; v -= 127)
285 if (
s->use_noise_coding) {
287 if (
s->channel_coded[ch]) {
289 n =
s->exponent_high_sizes[bsize];
290 for (
i = 0;
i < n;
i++) {
291 put_bits(&
s->pb, 1,
s->high_band_coded[ch][
i] = 0);
293 nb_coefs[ch] -=
s->exponent_high_bands[bsize][
i];
300 if (
s->block_len_bits !=
s->frame_len_bits)
305 if (
s->channel_coded[ch]) {
306 if (
s->use_exp_vlc) {
318 if (
s->channel_coded[ch]) {
321 tindex = (ch == 1 &&
s->ms_stereo);
322 ptr = &
s->coefs1[ch][0];
326 for (; ptr < eptr; ptr++) {
331 if (abs_level <= s->
coef_vlcs[tindex]->max_level)
332 if (run < s->
coef_vlcs[tindex]->levels[abs_level - 1])
333 code =
run +
s->int_table[tindex][abs_level - 1];
337 s->coef_vlcs[tindex]->huffcodes[
code]);
340 if (1 << coef_nb_bits <= abs_level)
343 put_bits(&
s->pb, coef_nb_bits, abs_level);
353 put_bits(&
s->pb,
s->coef_vlcs[tindex]->huffbits[1],
354 s->coef_vlcs[tindex]->huffcodes[1]);
363 uint8_t *buf,
int buf_size,
int total_gain)
367 if (
s->use_bit_reservoir)
383 s->block_len_bits =
s->frame_len_bits;
384 s->block_len = 1 <<
s->block_len_bits;
395 for (
i = 0;
i <
s->block_len;
i++) {
396 a =
s->coefs[0][
i] * 0.5;
397 b =
s->coefs[1][
i] * 0.5;
398 s->coefs[0][
i] =
a +
b;
399 s->coefs[1][
i] =
a -
b;
407 for (
i = 64;
i;
i >>= 1) {
414 while(total_gain <= 128 && error > 0)
417 av_log(avctx,
AV_LOG_ERROR,
"Invalid input data or requested bitrate too low, cannot encode\n");
438 #if CONFIG_WMAV1_ENCODER
454 #if CONFIG_WMAV2_ENCODER
static void error(const char *err)
int frame_size
Number of samples per channel in an audio frame.
@ AV_SAMPLE_FMT_FLTP
float, planar
static int nb_coefs(int length, int level, uint64_t sn)
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
static void align_put_bits(PutBitContext *s)
Pad the bitstream with zeros up to the next byte boundary.
static const CoefVLCTable coef_vlcs[6]
int sample_rate
samples per second
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
static int apply_window_and_mdct(AVCodecContext *avctx, const AVFrame *frame)
This structure describes decoded (raw) audio or video data.
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
static int put_bytes_count(const PutBitContext *s, int round_up)
static void init_exp(WMACodecContext *s, int ch, const int *exp_param)
int nb_channels
Number of channels in this layout.
av_cold int av_tx_init(AVTXContext **ctx, av_tx_fn *tx, enum AVTXType type, int inv, int len, const void *scale, uint64_t flags)
Initialize a transform context with the given configuration (i)MDCTs with an odd length are currently...
static float win(SuperEqualizerContext *s, float n, int N)
AVCodec p
The public AVCodec.
const struct AVCodec * codec
float WMACoef
type for decoded coefficients, int16_t would be enough for wma 1/2
AVChannelLayout ch_layout
Audio channel layout.
static int encode_block(WMACodecContext *s, float(*src_coefs)[BLOCK_MAX_SIZE], int total_gain)
int initial_padding
Audio only.
#define FF_CODEC_ENCODE_CB(func)
static av_cold int encode_init(AVCodecContext *avctx)
static int16_t mult(Float11 *f1, Float11 *f2)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
void(* av_tx_fn)(AVTXContext *s, void *out, void *in, ptrdiff_t stride)
Function pointer to a function to perform the transform.
@ AV_TX_FLOAT_MDCT
Standard MDCT with a sample data type of float, double or int32_t, respecively.
#define AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE
This encoder can reorder user opaque values from input AVFrames and return them with corresponding ou...
#define av_assert0(cond)
assert() equivalent, that is always enabled.
int ff_wma_total_gain_to_bits(int total_gain)
#define CODEC_LONG_NAME(str)
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
const FFCodec ff_wmav1_encoder
static int parse_exponents(DBEContext *s, DBEChannel *c)
static av_always_inline int64_t ff_samples_to_time_base(const AVCodecContext *avctx, int64_t samples)
Rescale from sample rate to AVCodecContext.time_base.
int64_t bit_rate
the average bitrate
const FFCodec ff_wmav2_encoder
static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
#define MAX_CODED_SUPERFRAME_SIZE
int ff_wma_end(AVCodecContext *avctx)
const uint8_t ff_aac_scalefactor_bits[121]
int(* init)(AVBSFContext *ctx)
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
#define AV_NOPTS_VALUE
Undefined timestamp value.
av_cold int ff_wma_init(AVCodecContext *avctx, int flags2)
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
static int fixed_exp(int x)
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
#define i(width, name, range_min, range_max)
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some it can consider them to be part of the FIFO and delay acknowledging a status change accordingly Example code
static int put_bits_count(PutBitContext *s)
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
AVSampleFormat
Audio sample formats.
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
const char * name
Name of the codec implementation.
static int encode_frame(WMACodecContext *s, float(*src_coefs)[BLOCK_MAX_SIZE], uint8_t *buf, int buf_size, int total_gain)
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
main external API structure.
static uint8_t * put_bits_ptr(PutBitContext *s)
Return the pointer to the byte where the bitstream writer will put the next bit.
static void encode_exp_vlc(WMACodecContext *s, int ch, const int *exp_param)
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
static void scale(int *out, const int *in, const int w, const int h, const int shift)
This structure stores compressed data.
int ff_alloc_packet(AVCodecContext *avctx, AVPacket *avpkt, int64_t size)
Check AVPacket size and allocate data.
const uint32_t ff_aac_scalefactor_code[121]